Telephones and other communications devices for sending audio information through the public switched telephone network (PSTN) have existed for a substantial period of time and are well understood. Recently, however, telephones have been created that exchange audio information through packet switched networks such as the Internet. These telephones use a digitized format that breaks down audio information into discrete Internet protocol (IP) packets. These voice over IP (VoIP) packets are sent and received through the packet switched network.
Exchanging audio information through a packet switched network presents a variety of problems. For example, errors in the packet transmission may result in echo, scratchy audio, dropped calls, delay and jitter in the audio string. These problems are compounded by the inability to debug intermittent voice quality issues in VoIP systems. Methods of notifying a repair staff are ineffective because users are unable to convey relevant information, such as the delay, and/or volume of an echo or the frequency of a buzz. Furthermore, often critical information is completely unavailable to the user, such as operating statistics. As long as the voice quality of VoIP is perceived to be inferior to the voice quality of traditional telephony systems, users will remain reluctant to employ VoIP systems.
Previous attempts to solve these problems include allowing a user to initiate a one-second recording of a call by depressing a certain series of buttons on a telephone keypad. This and other functionality using network-based resources was not completely effective in that the particular second of audio recorded may not have been the same audio in which the user was experiencing a problem or may not have provided an accurate representation of the audio the user was experiencing.